1. Queue - this application allows you to queue a call, in a call queue.

NOTE: This application is valid for Asterisk version 1.0.9 and above.

Syntax:

Queue(queuename[|options[|URL][|announceoverride][|timeout]])

List with the possible options

queuename - this is the name of the context in queues.conf, where you would like the call to be queued.options - you can use one or more of the following parameters

t - allow the called user to transfer the calling one to another numberT - allow the calling user to transfer the calld - data-quality (modem) call (minimum delay)h - allow the called person to hang up the line by pressing the asterisk key(*)H - allow the calling person to hang up the line by pressing the asterisk key(*)n - forbid the retries, if the timeout expires. It will exit and go to the next extension, instead.r - ring instead of playing music on hold.

URL - an URL will be sent to the called user, if the channel supports this.announceoverride - with this option you can set a sound file, which to replace the set one in the queues.conf file.timeout - the maximum time, in seconds, the call will wait in the queue. When this time expires, the next extension, by priority, will be executed. By default the timeout is set to 300 seconds.

In addition to the transferring of a call, it may be parked and then picked up by another user.

Purpose and usage

This application allows you to queue a phone call into a call queue. If somebody calls you, but the line is busy, because you already have another call, then the new caller will be put in the queue. The caller will hear music on hold until you are ready to answer its call.

To see how the application works, we recommend you to use our IAX softphone Idefisk. You can download it from here. Please also read our tutorial to learn how to configure it to work with Asterisk PBX.

Asterisk PBX configurations

NOTE: This is only an example of what for you can use this application. Of course you can use it and for other things.

iax.conf and sip.conf Configurations

We need one registered user in theiax.conf file and also another one in the sip.conf file. This is because we are going to use the IAX2 and the SIP channels. If you want to use other protocol such as MGCP, you have to do the configurations below respectively in mgcp.conf.

1) iax.conf

2)sip.conf

So, we have registered the user user1 in the iax.conf file and the user operator in the sip.conf file.

Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=test - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.

In the sip.conf file you can see the following option: disallow=all. This means that the line will not support any codecs. However, below this option we have allow=ulaw, allow=alaw and allow=gsm. This means that the line will support these three codecs - ulaw, alaw and gsm. It is important to write the options exactly in this order. First you write the disallow=all option and then the allow options. Otherwise, if you write the disallow option after the allow options, no codecs will be supported by the line.

queues.conf Configurations

In this file you have to say which users or agents will participate in your queue. In other words, which users or agents will be responsible for the answer of an incoming call. More about the features of this configuration file, you can learn in our tutorial about the queues.conf file.

There are two possible ways to configure the settings in this file.

The first one, is the so called "static" way. For this configuration you do not need the agents.conf file.

You will say directly, which users to answer the incoming call. In our example, we have created a context with the name [test]. In this context we have written the following: member => SIP/operator. This means that this user(operator) will be responsible for the answering of the incoming calls in the queue test. In the same way you can add even more than one user, which you would like to answer in this queue. However in our case we will have only one user. The advantage of this method, is that you do not have to use the AgentLogin or the AgentCallbackLogin applications in order to log the user in the queue. It will be logged in automatically, when it turns on its phone. The disadvantage is that this user can use, only the phone, on which it is registered. That is why the method is called "static". You can see a screenshot below.

The other way is the so called "dynamic" way. For this configuration you will need the agents.conf file.

In the section agents.conf Configurations we will show you the configurations in it and here we will explain you the configurations in queue.conf file.
Unlike the static method, here we have to write not the name of the user, but the number of the agent, as written in the agents.conf file. In our example we have written the following: member => Agent/8888. This means that we want the agent with number 8888, to answer the incoming calls in the queue test. If you decide to use this method you have to add in your dialplan either the AgentLogin or the AgentCallbackLogin application. This is necessary, because unlike the "static" method, here is not enough just to turn on your phone. The user has to log in itself in the queue, manually. The advantage of this method is that the user could log in from any place and phone, which is connected with the Asterisk PBX.You can see a screenshot below.

agents.conf Configurations

As we have mentioned above, we have to use this configuration file, if we decide to use the "dynamic" method. In this file, we have to create one or more agents. We will use them to answer the incoming calls in our queue.

The way of creating an agent is the following: agent => agentnumber,agentpassword,agentname. In our case we have written the following: agent => 8888,8888,operator. This means that we have registered an agent with number 8888, password 8888 and username operator. The agent is set in the [agents] context. The agent number will be used in the queues.conf file. Please, take a look at section queues.conf Configurations above, to learn how to do this.

Firstly, we need a logged in agent. For the purpose, we will use the AgentCallbackLogin application. The agent has to call the number 110, which will execute the AgentCallbackLogin application. As arguments in its brackets we have set the following: 8888,,@test. 8888 is the agent number. In this way, the system wonít ask us for the agentís number, it will ask us only for its password. We did not want to set any option, so we left the space between the commas blank. @test is the context in extensions.conf application, which contains the extension with the Dial application for the agent. When the agent log in, the system will ask it for its password and to enter an extension. The required extension is exactly the one with the Dial application. In our case the agent has to enter its password, which is 8888(set in the agents.conf file) and then the extension 111. This will cause the connection to the user operator through the SIP channel. Now the agent is ready to accept the incoming calls in our queue.

Now lets take a look at the Queue application. We have one extension, with number 112, which will execute the Queue application. As arguments in its brackets we have written the following: test|tThH|||100. Here test is the name of the context in the queues.conf file, where this agent is set. tT - this combination of letters will allow the called person and the caller to transfer the call. hH - this one will allow them to hang up the line by pressing the asterisk character(*). 100 is the timeout in seconds. If this time expires the call will be taken out from the queue and the next extension by priority, will be executed.

When somebody dials 112, the call will be attached in the queue. The caller will hear music on hold. Meanwhile the agentís phone will be ringing. When the agent answers the call, the music on hold will stop and the conversation will start. If in the same time somebody else dials 112, it wonít hear a busy signal, but again the music on hold. The new caller could stay on the line until it decides to hang up, the set timeout expires or the agent answers its call.

2. Screenshots of what you can see on the CLI of the Asterisk PBX

1) Agent log in

2) A call connected through the queue

3) Agent log off

3. Additional information

For more information about extensions.conf you can check here.
For more information about iax.conf you can check here.

This application is tested with our IAX softphone Idefisk. You can download it from here. For more information about this softphone please read our tutorial.

If you would like to test this application with the SIP channel you can read our tutorials about the SIP Softphones to learn how to configure them to work with Asterisk PBX

_____ (gabil4zglodek at mail dot ru)06 August 2018 16:57:06You can now easily login for a favorite online casino and click on the mobile tab. With all the betting sites and tipsters out there, there is no good reason that you should ever place a negative bet again. Money might be obtained just by alluding relatives on the web websites. https://wooricasino777.com

RobertIncew (owqn16925 at first dot baburn dot com)29 October 2017 18:35:57<a href=http://www.asianhairnskinclinic.com/images/class.php?p=91>Viagra Costo Attuale</a>
If you are looking to get started on a home business, investigation govt grants and personal loans available. A number of these support diverse areas of home business there is probably a single obtainable in the field you are planning to start-up in. Government permits and lending options might be a wonderful way to get start-up capital.
<a href=http://www.im.br/erros/lib.php?p=119>Kamagra Vente</a>
Working out regularly will work wonders to your levels of energy while you are coping with psoriatic joint disease. Try and strive for 15 to half an hour of exercise every day to provide a lot more energy and remain healthy as well as your bones solid. Be sure to not overdo it and find yourself tedious your self out.
<a href=http://www.mpcit.com.ar/mail/contact.php?jaaa=1-Apoteket-Ab-Viagra-Apoteket-Viagra-Pris-Viagra-F√∂r-Kvinnor-Effekt.html>Apoteket Ab Viagra</a>
Sportfishing is undoubtedly an encounter that can take a toll on your own body, while you must bring a cushion to rest on through the day. This will decrease the discomfort on your journey to help you increase the longevity of the sportfishing excursion. The better you feel the more energy you will get.
<a href=http://www.donadelli.ind.br/produtos/reduce.php?pa=Lovegra-Kaufen-Rezeptfrei,Viagra-F√ľr-M√§nner,Viagra-Generika-100mg-33>Lovegra Kaufen Rezeptfrei</a>

Eddie (or0zvlg46 at gmail dot com)20 December 2015 21:35:44on March 4, 2013 you are really a exlcleent webmaster. The site loading speed is incredible. It seems that you are doing any distinctive trick. Furthermore, The contents are masterpiece. you have performed a wonderful task in this subject! http://vmomda.com [url=http://rvymmfqjlj.com]rvymmfqjlj[/url] [link=http://snzbrqb.com]snzbrqb[/link]

Adeline (rpa67jt07e at mail dot com)18 December 2015 08:34:58Yay for sale season but my walelt is going to have to go into hibernation after this sale binge. Totally rocking the Saturday pants I scooped up from the J.Crew sale right. this. second, and I agree they are amazing!xx

vishu (vishugaddi at gmail dot com)23 July 2009 09:00:27how we will come to know that which queue member has answered the call so that we can proceed accordingly.

My requirement is, there are four members in a queue (named as SIP/101, 102, 103 ,104).
if Sip/101 answered the call then i want some action immebiatly, but if Sip/102 answered the call then some other action.

pls help, how to achieve it !!

Thanks in advance

suraj (suraj at linq dot in)25 July 2007 09:17:13Hello features.conf is working for normal sip user but not working for agents user...How to make ir work for them also..for normal user it takes transfer options from features.conf but not for agents user.Plss help me....

suraj (suraj at linq dot in)25 July 2007 07:47:36I am facing problem in transfering the call in within the local asterisk pbx..I am using grandstream phone...By default to transfer it takes hash button i dont know from where it is taking.......If I want to change transfer button.

TaiSHi (tujuanma at gmail dot com)13 March 2007 12:57:51Disable multiple lines on the Softphone.
That will do the trick.
Best regards.
Tai.

Jose Maria (sarachaga at yahoo dot com)11 May 2006 19:37:09Any hint on this issue? i have the same problem. I tried to limit the number of calls in sip.conf, but the user can't make another call ie. to transfer the call or make a conference.

GVC (gvc at gvcmedia dot dk)28 April 2006 15:51:27i have just one member of the type
member => SIP/operator
in my queue.

my problem is, when first caller calls, and is answered by the operator. Then 2nd caller calls, and now the queue tries to connect to the operator again (using sjphone i get another call to accept, and i get something like a conference if i accept)

why doesnt the queue see that the operator is allready busy answering a call?

this doesnt happen if i use agents, but agents and their log in routines seems a bit overkill when all i want is a queue for one very busy phone.

Darlon (darlon dot ferreira at betha dot com dot br)22 March 2006 18:02:14I made it in Asterisk, but I have a problem with idefisk. As the Idefisk have 6 lines, the softphone call in all 6 lines. Does have any way for enable only one call for softphone?

Thank you

Add Comment

Name:

Email:

Comment:

In order to prevent automatic posting on our website, we kindly request you to type in the number you see in the picture below.